1. Field of the Invention
The invention relates generally to network monitoring and, more particularly, to a device and method for monitoring a network at selected media segments.
2. Description of the Related Art
Real-time Transport Protocol (RTP), defined in Request for Comments (RFC) 3550, is widely used for the transmission of real-time or near-real-time data over packet networks. A Voice over Internet Protocol (VoIP) network may consist of one or more Internet Telephony Administrative Domains (ITADs), which include network components served by the same set of call route servers. An ITAD may be further broken down into geographic Points of Presence (POP), with each POP containing some number of gateways. Thus, an RTP media stream of a VoIP call may traverse multiple gateways to bridge up its calling and called parties.
RTP Control Protocol (RTCP) is a sister protocol of the RTP and provides out-of-band control information for an RTP media stream. A primary function of RTCP is to provide feedback on the quality of service (QoS) being provided by RTP. RTCP is used to monitor the media connection, collect statistics and information such as bytes sent, packets sent, lost packets, jitter, feedback and round trip delay, and periodically transmit control packets to participants in a streaming multimedia session (i.e., in the forward transmission direction) or as a feedback from a receiver back to a sender.
To ensure that IP networks meet customer expectations, service providers define Service Level Agreements and manage their networks to meet SLA requirements. Typically, performance measurements are taken from end-to-end (from the customer premise location). RTCP has been the prevailing approach to monitor its associated RTP stream for various voice metrics such as packet loss, inter-arrival jitter, round trip delay, etc. These voice metrics can be used at the endpoints (e.g., IP phones and originating/terminating voice gateways) of an RTP stream to judge the QoS or conformance check against an SLA from the perspective of these endpoints.
In general, a VoIP network may be composed of VoIP nodes bearing different ownership. Well-known VoIP nodes types include voice gateways, IP-to-IP gateways, and session border controllers (SBC). In a typical scenario, a backhaul VoIP service provider (VSP) sits between customer VoIP customer premise equipment (CPE) to bridge VoIP calls from one customer VoIP network to another. Thus, in many cases, a VSP does not control an entire call connection or session. Further, the backhaul VSP may either not have visibility to or may not be interested in looking into any VoIP node belonging to a customer's VoIP network.
Since there is no understanding of how each segment of a media path connecting the endpoints is performing with respect to voice quality, the existing end-to-end per call voice metric does not necessarily favor a VSP for justifying how well an SLA is being followed or locating a particular portion of media path that may be suffering a QoS issue.
Thus, there is a need to providing a fuller or more detailed picture of the media path QoS at a granular level while continuing to provide an end-to-end view of network performance.